A SIP Trunk is a logical connection between an IP PBX and a Service Provider’s application servers that allows voice over IP traffic to be exchanged between the two. When a call is placed from an internal phone to an external number, the PBX sends the necessary information to the SIP Trunk provider who establishes the call to the dialed number and acts as an intermediary for the call. All signaling and voice traffic between the PBX and the provider is exchanged using SIP and RTP protocol packets over the IP network.
If the number being called is a traditional PSTN telephone, the trunk provider routes the IP packets to a PSTN Gateway that is closest to the number being called, to minimize possible long distance charges. The provider can also terminate PSTN numbers, and route incoming calls for those numbers back to the IP PBX over the SIP can also Trunk. This allows businesses to offer local phone numbers in several geographical areas, but service them all from a single location.
If the number being called can be reached over a SIP Trunk, the call does not need to be routed over the PSTN, but can be carried on the IP network end to end.
Because a SIP Trunk is not a physical connection, there is no explicit limit on the number of calls that can be carried over a single trunk. Each call consumes a certain amount of network bandwidth, so the number of calls is limited by the amount of bandwidth that can flow between the IP PBX and the provider’s equipment.
